I have serious issues to get CME with SPA502g phones to work correctly. Incoming calls from outside are always routed to internal extension 204, then, they're xferred to some else internal phone (desired behavior). Sometimes, this call drops, but in debug voice ccapi inout log I don't see any reason. Only this line seems suspicious to me:
Sep 11 10:18:36.034: //1559/577A59248A27/CCAPI/ccCallDisconnect:
Cause Value=0, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
Cause value 0? Come on! Even google can't find explanation for that...
But this isn't the only problem i have .
When call is transferred internally, sound has terrible quality. When sound is transferred from outside world to any inside extension, no bad quality sound at all.
Also, transferred calls from outside world cannot be transfered again. Imagine situation like this: somebody calls from outside and has number xyzyyyxxx -> connected to 204 -> xfer to (i.e.) 211 -> wan't xfer to (i.e.) 205 but can't, phone doesn't offer that option. I am not sure if this is feature or failure, because that phone has only one line.
I collected this output from debug voice ccapi inout command whole afternoon and you can find this file as an attachment (it's quite large).
I am also attaching anonymized configuration. If you could see anything odd, please let me know.
Output from sh ver:
Cisco 2821 (revision 53.51) with 772096K/14336K bytes of memory.
System image file is "flash:c2800nm-adventerprisek9_ivs_li-mz.151-3.T4.bin"
Thanks for reply.
Then you are using the SPA phones as unsupported third-party SIP phones.
That is know to cause all kind of trouble and limitations. If you want good and solid results, only use 79xx or 69xx phones in SCCP mode.
Also, you should update IOS to latest 15.1(4)M4
could you please provide some additional informations about SPA502g not being supported as SIP phones with CME?
Whe had two types of phones: 7945 - SCCP (5 pieces) & 502g - SIP (40 pieces) - there was incompatibility when calling sccp -> sip and vice versa (not possible or call dropping). Because there are more SIP than SCCP phones, we decided to keep SIP phones and abandon SCCP. Also, there were issues when calling from SIP trunk to SCCP. For a few days, this configuration was running properly. Seems to me like a mystery why this is failing again. This project is giving me headache for a few weeks and I need to know if there is any way how to solve it or with this configuration and phones I can't achieve valuable results.
I'll also try to upgrade to latest IOS, but I have serious doubts and I think it won't change anything.
Thank you for your reply and have a nice day.
Under voice register pool, enter type ?
That will show you the supported SIP phones.
However CME works good only with SCCP phones. If you really want to use SPA phones you best choice would be to use an UC500 that supports them in SPCP mode, that is similar to SCCP.
Hmm, tough situation. SPA502g aren't supported by CME and few weeks ago, I read somewhere that SIP firmware in 7945 is error-prone, some versions are very buggy and this configuration is not supported by cisco. However, I am going to revise every command in my configuration once more and I hope at least calls xferring will work fine.
However, do you think there could be problem in fact, that phones are registering to WAN interface of CME? I remember I had good reason for that when I was trying to get SCCP and SIP phones to work together....
Thanks, have a nice day.
What is the modality configured on your CME to handle xfer?
You can try these commands:
voice service voip
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
In this case the CME acts as B2B user agent and a new SIP INVITE is generated in case of xfer.
I am afraid I don't truly understand the "modality" term. But if it means what method (hairpin or h450.2) is my CME using, then i guess it's h450.2 because it isn't explicitly disabled on CME.
I already have no supplementary-service sip refer in my config, I can add "sip moved-temporarily". But I don't see how it could influence present state.
But as I went through my config again and again, I saw there is some config influencing behavior of SDP (session description protocol I guess?) in outgoing dial-peers (1,4,6) but there isn't any config in incoming dial-peer (2). I mean this commands:
no voice-class sip pass-thru headers
voice-class sip pass-thru content unsupp
voice-class sip pass-thru content sdp
Do you think it could be reason of my biggest issue? (That is- only some xferred calls from outside are dropped after some time, but some aren't ....That's what is driving me crazy, crazy, crazy. I caught debug from debug ccapi voice inout during whole one business day again and I can upload it if you want take a look at it - I don't see anything suspicious).
Thank you all in advance.
PS: Sorry for maybe stupid questions, but I am newbie in a voice world and this CME config is part of project that is very delayed and should be finished ASAP.
H.450 describes a family of supplementary services of H.323 signalling protocol.
In your config, dial-peers are configured to use SIP. Also the SPA phone uses SIP. So H.450 is not used.
Can you add a "debug ccsip all" instead of "voice ccapi" of a single xferred failed call?
Thank you for clarification, I need to improve my knowledge in VoIP. First of all, I'd like to ask you to recommend some good book about CME and VoIP.
About output from debug ccsip all:
Yes, i will provide it tomorrow at about 4 p.m. CET. The network, where CME resides, is production network where users are already working and there's no way for me to be onsite during regular business hours. Of course there could be some SIP messages from more calls mixed, but I hope I'll know rough time when call was dropped and participating phone numbers.
Thank you for your effort,
Have a nice evening
Good evening Daniele,
Sorry I didn't provide any output yesterday - there was no call dropped, therefore I didn't have anything to provide.
But another remark came to my mind. Technician, who is taking care of network on-site told me he has suspicion only those calls with incoming numbers having first three numbers same as internal numbers are being dropped.
e.g. There is an incoming number beginning with 224 from outside telephone network and we have also internal phone with number 224. Do you think it could be the reason?
Good evening again,
still having issues with bad quality and dropping calls when transferred, as mentioned above. There is an attachment showing SIP messages flowing between our CUCME and SIP gateway. Unfortunately, that call wasn't dropped, that call was hung up correctly. Log was anonymized in order to hide calling and called number, instead of it - first six digits of both numbers were substituted with letters.
Number abcde820 was calling number and number tuvxyz934 was called number. From internal point of view, called number was xlated to extension 204 and that call was transferred to extension 211.
There are few errors at the end of the log (If you are going to read it, I'd suggest to start at the end), but do not understand what is wrong and why those errors appeared. Also, I don't understand why there is "100 Trying" at line 69209 - it's at time when call was already established. If somebody could clarify it to me, I'd be happy.
Too many logs.
At the beginning of logs I see an XFER with REFER but not others with this method. I don't see CME act as sip to sip agent (B2BUA).
I suggest you these checks:
- disable audio codecs not used on phones and CME
You have configured G711u (codec 0) but your calls use G711a (codec 8). During setup your devices negotiate some codecs:
codec: 18 99 8 0
200 OK Cisco/SPA502G-7.5.3
codec 8 99
codec 8 0 101
codec 8 101 0 ## something is wrong
coedc 8 0 2 9 18 96 97 98 101
SPA502 codec config:
CME codec config:
disable SDP passthrough and use CME as B2B
voice service voip
no pass-thru content unsupp
no pass-thru content sdp
dial-peer voice 1-4-6
no voice-class sip pass-thru content unsupp
no voice-class sip pass-thru content sdp
In this way your calls will use g711u as configured in voice registers and dial-peers.
- disable SIP REFER on SPA502
- upgrade SPA502 firmware with 7.5.5
actually you have Cisco/SPA502G-7.5.3